<div dir="ltr"><div><div><div>Ok I found how to remove the loopback but still having the same issue.<br><br></div>I noticed that after I removed the loopback the hold music that was playing before the voicemail connect was also of poor quality if that means anything? My dialplan now connects the call straight to voicemail so I don't hear ringback or anything but the issue remains.<br>
<br></div>Interestingly the speech breaks up at the same point of the announcement every time, I hear "The person at extension ?000 is ????available, ???? your message at the tone, press any key or stop talking to end the recording. Then some weird digital noise after the beep.<br>
<br></div>Hope this is of some use<br></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Mar 19, 2013 at 1:55 PM, Elliot Farmer <span dir="ltr"><<a href="mailto:elliotfarmer101@gmail.com" target="_blank">elliotfarmer101@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>Sorry I'm pretty new to FreeSwitch, how would I go about that? I'm using the default config that has loopback setup as the route to voicemail as far as I can work out.<br>
<br></div>Is there a way of sending to voicemail without a loopback?<div><div class="h5"><br>
<div><div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Mar 19, 2013 at 1:10 PM, Yuriy Nasida <span dir="ltr"><<a href="mailto:nasida@live.ru" target="_blank">nasida@live.ru</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><div dir="ltr"><div><font face="sans-serif"><span style="line-height:19.046875px">looked at your logs. Do you use loopback endpoint in your dialplan ? If yes, can you check without this ?</span></font></div><br><div>
<div></div><hr>Date: Tue, 19 Mar 2013 12:37:47 +0000<br>From: <a href="mailto:elliotfarmer101@gmail.com" target="_blank">elliotfarmer101@gmail.com</a><br>To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
Subject: Re: [Freeswitch-users] External voicemail issue<div><div><br><br><div dir="ltr"><div><div><div>I've done some further testing and the issue appears to be isolated to the voicemail system.<br><br></div>
To test I configured the inbound dialplan to play two wav files before transferring the call...<br>
<br>1) 8kz wav from wikipedia <a href="http://www.nch.com.au/acm/8kulaw.wav" target="_blank">http://www.nch.com.au/acm/8kulaw.wav</a><br>2) /usr/local/freeswitch/sounds/en/us/callie/ivr/48000/ivr-you_are_number_one.wav<br>
<br></div>Both play fine, as soon as the call is transferred to voicemail (via extension 1000 that is not registered at the time) the quality drops and I start loosing speech. I've tried preferring the PCMU codec within vars but I seem to have a lot of
"TRANSCODING_NECESSARY" messages, they are not isolated to the
voicemail steps however.<br><br></div>Has anyone got an idea where I can start to look? Any help would be much appreciated.<br><div><div><div><div><div><div><div><div><div><div><div><div><div><div><div>
<br><br><div>On Mon, Mar 18, 2013 at 7:48 PM, Elliot Farmer <span dir="ltr"><<a href="mailto:elliotfarmer101@gmail.com" target="_blank">elliotfarmer101@gmail.com</a>></span> wrote:<br><blockquote style="border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div dir="ltr"><div><div><div><div><div><div>Hi all,<br><br></div>I'm having an issue with the voicemail system when accessing via a Skype connect trunk. <br><br></div>Internally all is working, however if I call my external Skype number from a PSTN line when the call is routed to voicemail I get a lot of intermittent speech loss during the IVR announcement and then after the beep I can hear odd sounds that I can only describe as digital interference type sounds. Also the IVR is very quick to say "the recording is below the minimum length" although if I speak quickly I can record a voicemail and it sounds fine during playback, not amazing audio quality but no major issues.<br>
<br></div>The issue seems to be the same as <a href="http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066258.html" target="_blank">http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066258.html</a>, I have tried recording a greeting but the same issues occur.<br>
<br></div>I'm running Freeswitch on CentOS 6.3 on a physical machine.<br><br></div>Here are some logs <a href="http://pastebin.com/jmuks47q" target="_blank">http://pastebin.com/jmuks47q</a> , I've tried to remove the phone numbers and IP addresses as I don't have permission from the owners to publish them on the internet.<br>
<br></div>Thanks in advance for your help! <br><div><div><div><br><br></div></div></div></div>
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